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ATA191-3PW-K9 is the MPP version. The ATA191-K9 version is the one that works with Cisco firmware. Either one should work with CUCM. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

The username and password would be provided by Callwithus. Its how they are authenticating your SIP trunk to them. So the SIP UA stuff is copy and paste and change the username and password. The dial peer portion would be more in line with your organization. Meaning it would look something more...

I am not 100% sure if creating a Translation Pattern and updating the mask would work. Why not use SNR though? Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

Demo licenses would have an expiration on them that Cisco can give you to hold you over. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

My first thought would be a codec mismatch. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

So in that case i would do the TOD schedule in CUCM to send to the appropriate call handler. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

Why not have 1 CTIRP pointing to the call handler and then set the time of day scheduling there with your greetings? Instead of 2. Granted its been years since ive done TOD stuff with Unity. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

More than likely the port config on the router would be the place to look. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

Itll be controlled by your telco, then technically by your licensing. So easiest thing to do is ask your telco what its set for to start with. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

Generally it would mean your CSS isnt correct. It would need to be whatever you use to dial externally. Also if you use a prefix, make sure thats in there somewhere too. So the TP would look something like this: Pattern = The extension of the person. Partition = Whatever partition you put your...

You can do it a few ways but i typically do it via the Translation Pattern. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

I have hundreds of Valcom's hooked up to FXS ports without issue. Though i use a 9970...now its a 9972 i believe. But anyhow, sounds to me like the dipswitches on the Valcom arent correct. Youd have to reference the manual and maybe play around with it even to get it correct. But they do work...

...For example: I have something like 950 locations. Each location has 1 partition. There is 4 digit overlap. Each location can dial another via *XXXX which is the location number. Can they specifically dial another phone at another location that does not have a 10 digit number? No. But they...

How are you dialing the other locations? 4 digits? I assume each location has overlap in the extensions and thats why you have so many partitions? Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

I would definitely say you have a routing problem of some sort. Whether a firewall is blocking it or just a hop doesnt have a route it needs. For ports i would start with 5060/5061 for SIP and then 16384 to 32767 is the range for RTP. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

First guess would be the VPN Firewall is blocking SIP traffic. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

What audible message do you get when trying to place an outbound call? Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

Then I would say, its not routing the way you think it is. Meaning something else is in play. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

That would be the way to increase it. Rings are 4 seconds so if you wanted it for, i would set it for 16. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice

Any reason you are using FXO vs FXS? I use FXS ports without issue. Certifications: A+ Network+ CCENT CCNA Voice CCNP Voice